Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
index 252ffb2a3e59737e125e10c605b137ec9d67aeca..9d134afbccc81a934c0146af9ffb116395dc1ac1 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -695,6 +695,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) { |
size_t length = IP_PACKET_SIZE; |
uint8_t data_buffer[IP_PACKET_SIZE]; |
int64_t capture_time_ms; |
+ |
if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true, |
data_buffer, &length, |
&capture_time_ms)) { |
@@ -922,8 +923,8 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, |
// TODO(sprang): Potentially too much overhead in IsRegistered()? |
bool using_transport_seq = rtp_header_extension_map_.IsRegistered( |
kRtpExtensionTransportSequenceNumber) && |
- transport_sequence_number_allocator_ && |
- !is_retransmit; |
+ transport_sequence_number_allocator_; |
+ |
PacketOptions options; |
if (using_transport_seq) { |
options.packet_id = |